Module webrtc

The webrtc module contains WebRTC integrations.

Summary

Members Descriptions
namespacescy

namespace scy

Summary

Members Descriptions
classscy::AudioPacketModule
classscy::DummySetSessionDescriptionObserver
classscy::ImageSequenceRecorder
classscy::MultiplexMediaCapturer
classscy::PeerConnection
classscy::PeerConnectionManager
classscy::StreamRecorder
classscy::VideoPacketSource

class scy::AudioPacketModule

class scy::AudioPacketModule
  : public AudioDeviceModule
  : public MessageHandler

This class implements an AudioDeviceModule that can be used to detect if audio is being received properly if it is fed by another AudioDeviceModule in some arbitrary audio pipeline where they are connected. It does not play out or record any audio so it does not need access to any hardware and can therefore be used in the gtest testing framework.

Note P postfix of a function indicates that it should only be called by the processing thread.

Summary

Members Descriptions
public void onAudioCaptured(void * sender,av::AudioPacket& packet) Handles input packets from the capture for sending.
public int64_t TimeUntilNextProcess()
public void Process()
public int32_t ActiveAudioLayer(AudioLayer * audio_layer) const
public ErrorCode LastError() const
public int32_t RegisterEventObserver(webrtc::AudioDeviceObserver * event_callback)
public int32_t RegisterAudioCallback(webrtc::AudioTransport * audio_callback) Note: Calling this method from a callback may result in deadlock.
public int32_t Init()
public int32_t Terminate()
public bool Initialized() const
public int16_t PlayoutDevices()
public int16_t RecordingDevices()
public int32_t PlayoutDeviceName(uint16_t index,char name,char guid)
public int32_t RecordingDeviceName(uint16_t index,char name,char guid)
public int32_t SetPlayoutDevice(uint16_t index)
public int32_t SetPlayoutDevice(WindowsDeviceType device)
public int32_t SetRecordingDevice(uint16_t index)
public int32_t SetRecordingDevice(WindowsDeviceType device)
public int32_t PlayoutIsAvailable(bool * available)
public int32_t InitPlayout()
public bool PlayoutIsInitialized() const
public int32_t RecordingIsAvailable(bool * available)
public int32_t InitRecording()
public bool RecordingIsInitialized() const
public int32_t StartPlayout()
public int32_t StopPlayout()
public bool Playing() const
public int32_t StartRecording()
public int32_t StopRecording()
public bool Recording() const
public int32_t SetAGC(bool enable)
public bool AGC() const
public int32_t SetWaveOutVolume(uint16_t volume_left,uint16_t volume_right)
public int32_t WaveOutVolume(uint16_t * volume_left,uint16_t * volume_right) const
public int32_t InitSpeaker()
public bool SpeakerIsInitialized() const
public int32_t InitMicrophone()
public bool MicrophoneIsInitialized() const
public int32_t SpeakerVolumeIsAvailable(bool * available)
public int32_t SetSpeakerVolume(uint32_t volume)
public int32_t SpeakerVolume(uint32_t * volume) const
public int32_t MaxSpeakerVolume(uint32_t * max_volume) const
public int32_t MinSpeakerVolume(uint32_t * min_volume) const
public int32_t SpeakerVolumeStepSize(uint16_t * step_size) const
public int32_t MicrophoneVolumeIsAvailable(bool * available)
public int32_t SetMicrophoneVolume(uint32_t volume)
public int32_t MicrophoneVolume(uint32_t * volume) const
public int32_t MaxMicrophoneVolume(uint32_t * max_volume) const
public int32_t MinMicrophoneVolume(uint32_t * min_volume) const
public int32_t MicrophoneVolumeStepSize(uint16_t * step_size) const
public int32_t SpeakerMuteIsAvailable(bool * available)
public int32_t SetSpeakerMute(bool enable)
public int32_t SpeakerMute(bool * enabled) const
public int32_t MicrophoneMuteIsAvailable(bool * available)
public int32_t SetMicrophoneMute(bool enable)
public int32_t MicrophoneMute(bool * enabled) const
public int32_t MicrophoneBoostIsAvailable(bool * available)
public int32_t SetMicrophoneBoost(bool enable)
public int32_t MicrophoneBoost(bool * enabled) const
public int32_t StereoPlayoutIsAvailable(bool * available) const
public int32_t SetStereoPlayout(bool enable)
public int32_t StereoPlayout(bool * enabled) const
public int32_t StereoRecordingIsAvailable(bool * available) const
public int32_t SetStereoRecording(bool enable)
public int32_t StereoRecording(bool * enabled) const
public int32_t SetRecordingChannel(const ChannelType channel)
public int32_t RecordingChannel(ChannelType * channel) const
public int32_t SetPlayoutBuffer(const BufferType type,uint16_t size_ms)
public int32_t PlayoutBuffer(BufferType * type,uint16_t * size_ms) const
public int32_t PlayoutDelay(uint16_t * delay_ms) const
public int32_t RecordingDelay(uint16_t * delay_ms) const
public int32_t CPULoad(uint16_t * load) const
public int32_t StartRawOutputFileRecording(const char pcm_file_name_utf8)
public int32_t StopRawOutputFileRecording()
public int32_t StartRawInputFileRecording(const char pcm_file_name_utf8)
public int32_t StopRawInputFileRecording()
public int32_t SetRecordingSampleRate(const uint32_t samples_per_sec)
public int32_t RecordingSampleRate(uint32_t * samples_per_sec) const
public int32_t SetPlayoutSampleRate(const uint32_t samples_per_sec)
public int32_t PlayoutSampleRate(uint32_t * samples_per_sec) const
public int32_t ResetAudioDevice()
public int32_t SetLoudspeakerStatus(bool enable)
public int32_t GetLoudspeakerStatus(bool * enabled) const
public inline bool BuiltInAECIsAvailable() const
public inline int32_t EnableBuiltInAEC(bool enable)
public inline bool BuiltInAGCIsAvailable() const
public inline int32_t EnableBuiltInAGC(bool enable)
public inline bool BuiltInNSIsAvailable() const
public inline int32_t EnableBuiltInNS(bool enable)
public void OnMessage(rtc::Message * msg) WEBRTC_IOS.
protected explicit AudioPacketModule()
protected virtual ~AudioPacketModule()

Members

public void onAudioCaptured(void * sender,av::AudioPacket& packet)

Handles input packets from the capture for sending.

public int64_t TimeUntilNextProcess()

Following functions are inherited from webrtc::AudioDeviceModule. Only functions called by [PeerConnection](#classscy_1_1PeerConnection) are implemented, the rest do nothing and return success. If a function is not expected to be called by [PeerConnection](#classscy_1_1PeerConnection) an assertion is triggered if it is in fact called.

public void Process()

public int32_t ActiveAudioLayer(AudioLayer * audio_layer) const

public ErrorCode LastError() const

public int32_t RegisterEventObserver(webrtc::AudioDeviceObserver * event_callback)

public int32_t RegisterAudioCallback(webrtc::AudioTransport * audio_callback)

Note: Calling this method from a callback may result in deadlock.

public int32_t Init()

public int32_t Terminate()

public bool Initialized() const

public int16_t PlayoutDevices()

public int16_t RecordingDevices()

public int32_t PlayoutDeviceName(uint16_t index,char name,char guid)

public int32_t RecordingDeviceName(uint16_t index,char name,char guid)

public int32_t SetPlayoutDevice(uint16_t index)

public int32_t SetPlayoutDevice(WindowsDeviceType device)

public int32_t SetRecordingDevice(uint16_t index)

public int32_t SetRecordingDevice(WindowsDeviceType device)

public int32_t PlayoutIsAvailable(bool * available)

public int32_t InitPlayout()

public bool PlayoutIsInitialized() const

public int32_t RecordingIsAvailable(bool * available)

public int32_t InitRecording()

public bool RecordingIsInitialized() const

public int32_t StartPlayout()

public int32_t StopPlayout()

public bool Playing() const

public int32_t StartRecording()

public int32_t StopRecording()

public bool Recording() const

public int32_t SetAGC(bool enable)

public bool AGC() const

public int32_t SetWaveOutVolume(uint16_t volume_left,uint16_t volume_right)

public int32_t WaveOutVolume(uint16_t * volume_left,uint16_t * volume_right) const

public int32_t InitSpeaker()

public bool SpeakerIsInitialized() const

public int32_t InitMicrophone()

public bool MicrophoneIsInitialized() const

public int32_t SpeakerVolumeIsAvailable(bool * available)

public int32_t SetSpeakerVolume(uint32_t volume)

public int32_t SpeakerVolume(uint32_t * volume) const

public int32_t MaxSpeakerVolume(uint32_t * max_volume) const

public int32_t MinSpeakerVolume(uint32_t * min_volume) const

public int32_t SpeakerVolumeStepSize(uint16_t * step_size) const

public int32_t MicrophoneVolumeIsAvailable(bool * available)

public int32_t SetMicrophoneVolume(uint32_t volume)

public int32_t MicrophoneVolume(uint32_t * volume) const

public int32_t MaxMicrophoneVolume(uint32_t * max_volume) const

public int32_t MinMicrophoneVolume(uint32_t * min_volume) const

public int32_t MicrophoneVolumeStepSize(uint16_t * step_size) const

public int32_t SpeakerMuteIsAvailable(bool * available)

public int32_t SetSpeakerMute(bool enable)

public int32_t SpeakerMute(bool * enabled) const

public int32_t MicrophoneMuteIsAvailable(bool * available)

public int32_t SetMicrophoneMute(bool enable)

public int32_t MicrophoneMute(bool * enabled) const

public int32_t MicrophoneBoostIsAvailable(bool * available)

public int32_t SetMicrophoneBoost(bool enable)

public int32_t MicrophoneBoost(bool * enabled) const

public int32_t StereoPlayoutIsAvailable(bool * available) const

public int32_t SetStereoPlayout(bool enable)

public int32_t StereoPlayout(bool * enabled) const

public int32_t StereoRecordingIsAvailable(bool * available) const

public int32_t SetStereoRecording(bool enable)

public int32_t StereoRecording(bool * enabled) const

public int32_t SetRecordingChannel(const ChannelType channel)

public int32_t RecordingChannel(ChannelType * channel) const

public int32_t SetPlayoutBuffer(const BufferType type,uint16_t size_ms)

public int32_t PlayoutBuffer(BufferType * type,uint16_t * size_ms) const

public int32_t PlayoutDelay(uint16_t * delay_ms) const

public int32_t RecordingDelay(uint16_t * delay_ms) const

public int32_t CPULoad(uint16_t * load) const

public int32_t StartRawOutputFileRecording(const char pcm_file_name_utf8)

public int32_t StopRawOutputFileRecording()

public int32_t StartRawInputFileRecording(const char pcm_file_name_utf8)

public int32_t StopRawInputFileRecording()

public int32_t SetRecordingSampleRate(const uint32_t samples_per_sec)

public int32_t RecordingSampleRate(uint32_t * samples_per_sec) const

public int32_t SetPlayoutSampleRate(const uint32_t samples_per_sec)

public int32_t PlayoutSampleRate(uint32_t * samples_per_sec) const

public int32_t ResetAudioDevice()

public int32_t SetLoudspeakerStatus(bool enable)

public int32_t GetLoudspeakerStatus(bool * enabled) const

public inline bool BuiltInAECIsAvailable() const

public inline int32_t EnableBuiltInAEC(bool enable)

public inline bool BuiltInAGCIsAvailable() const

public inline int32_t EnableBuiltInAGC(bool enable)

public inline bool BuiltInNSIsAvailable() const

public inline int32_t EnableBuiltInNS(bool enable)

public void OnMessage(rtc::Message * msg)

WEBRTC_IOS.

End of functions inherited from webrtc::AudioDeviceModule. The following function is inherited from rtc::MessageHandler.

protected explicit AudioPacketModule()

The constructor is protected because the class needs to be created as a reference counted object (for memory managment reasons). It could be exposed in which case the burden of proper instantiation would be put on the creator of a AudioPacketModule instance. To create an instance of this class use the [Create()](#group__webrtc_1gaf78f45016b72c9bdca424e9a31691db8) API.

protected virtual ~AudioPacketModule()

The destructor is protected because it is reference counted and should not be deleted directly.

class scy::DummySetSessionDescriptionObserver

class scy::DummySetSessionDescriptionObserver
  : public SetSessionDescriptionObserver

Summary

Members Descriptions
public virtual void OnSuccess()
public virtual void OnFailure(const std::string & error)
protected inline DummySetSessionDescriptionObserver()
protected inline ~DummySetSessionDescriptionObserver()

Members

public virtual void OnSuccess()

public virtual void OnFailure(const std::string & error)

protected inline DummySetSessionDescriptionObserver()

protected inline ~DummySetSessionDescriptionObserver()

class scy::ImageSequenceRecorder

class scy::ImageSequenceRecorder
  : public rtc::VideoSinkInterface< cricket::VideoFrame >

Summary

Members Descriptions
public ImageSequenceRecorder(webrtc::VideoTrackInterface * track_to_render,const std::string & basename)
public virtual ~ImageSequenceRecorder()
public std::string getNextFilename()
public void OnFrame(const cricket::VideoFrame & frame) VideoSinkInterface implementation.
protected rtc::scoped_refptr< webrtc::VideoTrackInterface > _renderedTrack
protected const std::string _basename
protected size_t _count
protected int _width
protected int _height
protected av::VideoEncoder _encoder
protected AVFrame * _avframe

Members

public ImageSequenceRecorder(webrtc::VideoTrackInterface * track_to_render,const std::string & basename)

public virtual ~ImageSequenceRecorder()

public std::string getNextFilename()

public void OnFrame(const cricket::VideoFrame & frame)

VideoSinkInterface implementation.

protected rtc::scoped_refptr< webrtc::VideoTrackInterface > _renderedTrack

protected const std::string _basename

protected size_t _count

protected int _width

protected int _height

protected av::VideoEncoder _encoder

protected AVFrame * _avframe

class scy::MultiplexMediaCapturer

Summary

Members Descriptions
public MultiplexMediaCapturer()
public virtual ~MultiplexMediaCapturer()
public virtual void openFile(const std::string & file)
public virtual void addMediaTracks(webrtc::PeerConnectionFactoryInterface * factory,webrtc::MediaStreamInterface * stream)
public virtual void start()
public virtual void stop()
public virtual rtc::scoped_refptr<AudioPacketModule> getAudioModule()
public virtualVideoPacketSource* createVideoSource()
protectedPacketStream_stream
protected av::MediaCapture::Ptr _capture
protected rtc::scoped_refptr<AudioPacketModule> _audioModule

Members

public MultiplexMediaCapturer()

public virtual ~MultiplexMediaCapturer()

public virtual void openFile(const std::string & file)

public virtual void addMediaTracks(webrtc::PeerConnectionFactoryInterface * factory,webrtc::MediaStreamInterface * stream)

public virtual void start()

public virtual void stop()

public virtual rtc::scoped_refptr<AudioPacketModule> getAudioModule()

public virtualVideoPacketSource* createVideoSource()

protectedPacketStream_stream

protected av::MediaCapture::Ptr _capture

protected rtc::scoped_refptr<AudioPacketModule> _audioModule

class scy::PeerConnection

class scy::PeerConnection
  : public PeerConnectionObserver
  : public CreateSessionDescriptionObserver

Summary

Members Descriptions
public PeerConnection(PeerConnectionManager* manager,const std::string & peerid,Modemode)
public virtual ~PeerConnection()
public virtual rtc::scoped_refptr< webrtc::MediaStreamInterface > createMediaStream()
public virtual void createConnection()
public virtual void closeConnection() Close the peer connection.
public virtual void createOffer()
public virtual void recvSDP(const std::string & type,const std::string & sdp) Receive a remote offer or answer.
public virtual void recvCandidate(const std::string & mid,int mlineindex,const std::string & sdp) Receive a remote candidate.
public void setPeerConnectionFactory(rtc::scoped_refptr< webrtc::PeerConnectionFactoryInterface > factory)
public std::string peerid() const
public webrtc::FakeConstraints & constraints()
public webrtc::PeerConnectionFactoryInterface * factory() const
public rtc::scoped_refptr< webrtc::PeerConnectionInterface > peerConnection() const
public rtc::scoped_refptr< webrtc::MediaStreamInterface > stream() const
protectedPeerConnectionManager* _manager
protected std::string _peerid
protectedMode_mode
protected webrtc::PeerConnectionInterface::RTCConfiguration _config
protected webrtc::FakeConstraints _constraints
protected rtc::scoped_refptr< webrtc::PeerConnectionFactoryInterface > _factory
protected rtc::scoped_refptr< webrtc::PeerConnectionInterface > _peerConnection
protected rtc::scoped_refptr< webrtc::MediaStreamInterface > _stream
protected virtual void OnAddStream(webrtc::MediaStreamInterface * stream) inherited from PeerConnectionObserver
protected virtual void OnRemoveStream(webrtc::MediaStreamInterface * stream)
protected virtual void OnIceCandidate(const webrtc::IceCandidateInterface * candidate)
protected virtual void OnSignalingChange(webrtc::PeerConnectionInterface::SignalingState new_state)
protected virtual void OnIceConnectionChange(webrtc::PeerConnectionInterface::IceConnectionState new_state)
protected virtual void OnIceGatheringChange(webrtc::PeerConnectionInterface::IceGatheringState new_state)
protected virtual void OnRenegotiationNeeded()
protected virtual void OnSuccess(webrtc::SessionDescriptionInterface * desc) inherited from CreateSessionDescriptionObserver
protected virtual void OnFailure(const std::string & error)
protected inline virtual int AddRef() const
protected inline virtual int Release() const

Members

public PeerConnection(PeerConnectionManager* manager,const std::string & peerid,Modemode)

public virtual ~PeerConnection()

public virtual rtc::scoped_refptr< webrtc::MediaStreamInterface > createMediaStream()

Create the local media stream. Only necessary when we are creating the offer.

public virtual void createConnection()

Create the peer connection once configuration, constraints and streams have been set.

public virtual void closeConnection()

Close the peer connection.

public virtual void createOffer()

Create the offer SDP tos end to the peer. No offer should be received after creating the offer. A call to recvRemoteAnswer is expected to initiate the session.

public virtual void recvSDP(const std::string & type,const std::string & sdp)

Receive a remote offer or answer.

public virtual void recvCandidate(const std::string & mid,int mlineindex,const std::string & sdp)

Receive a remote candidate.

public void setPeerConnectionFactory(rtc::scoped_refptr< webrtc::PeerConnectionFactoryInterface > factory)

Set a custom PeerConnectionFactory object Must be done before any streams are initiated

public std::string peerid() const

public webrtc::FakeConstraints & constraints()

public webrtc::PeerConnectionFactoryInterface * factory() const

public rtc::scoped_refptr< webrtc::PeerConnectionInterface > peerConnection() const

public rtc::scoped_refptr< webrtc::MediaStreamInterface > stream() const

protectedPeerConnectionManager* _manager

protected std::string _peerid

protectedMode_mode

protected webrtc::PeerConnectionInterface::RTCConfiguration _config

protected webrtc::FakeConstraints _constraints

protected rtc::scoped_refptr< webrtc::PeerConnectionFactoryInterface > _factory

protected rtc::scoped_refptr< webrtc::PeerConnectionInterface > _peerConnection

protected rtc::scoped_refptr< webrtc::MediaStreamInterface > _stream

protected virtual void OnAddStream(webrtc::MediaStreamInterface * stream)

inherited from PeerConnectionObserver

protected virtual void OnRemoveStream(webrtc::MediaStreamInterface * stream)

protected virtual void OnIceCandidate(const webrtc::IceCandidateInterface * candidate)

protected virtual void OnSignalingChange(webrtc::PeerConnectionInterface::SignalingState new_state)

protected virtual void OnIceConnectionChange(webrtc::PeerConnectionInterface::IceConnectionState new_state)

protected virtual void OnIceGatheringChange(webrtc::PeerConnectionInterface::IceGatheringState new_state)

protected virtual void OnRenegotiationNeeded()

protected virtual void OnSuccess(webrtc::SessionDescriptionInterface * desc)

inherited from CreateSessionDescriptionObserver

protected virtual void OnFailure(const std::string & error)

protected inline virtual int AddRef() const

protected inline virtual int Release() const

class scy::PeerConnectionManager

class scy::PeerConnectionManager
  : public scy::PointerCollection< std::string, PeerConnection >

Summary

Members Descriptions
public PeerConnectionManager(rtc::scoped_refptr< webrtc::PeerConnectionFactoryInterface > factory)
public virtual ~PeerConnectionManager()
public virtual void sendSDP(PeerConnection* conn,const std::string & type,const std::string & sdp)
public virtual void sendCandidate(PeerConnection* conn,const std::string & mid,int mlineindex,const std::string & sdp)
public virtual void recvSDP(const std::string & peerid,const json::Value & data)
public virtual void recvCandidate(const std::string & peerid,const json::Value & data)
public virtual void onAddRemoteStream(PeerConnection* conn,webrtc::MediaStreamInterface * stream)
public virtual void onRemoveRemoteStream(PeerConnection* conn,webrtc::MediaStreamInterface * stream)
public virtual void onStable(PeerConnection* conn)
public virtual void onClosed(PeerConnection* conn)
public virtual void onFailure(PeerConnection* conn,const std::string & error)
public webrtc::PeerConnectionFactoryInterface * factory() const
protected rtc::scoped_refptr< webrtc::PeerConnectionFactoryInterface > _factory

Members

public PeerConnectionManager(rtc::scoped_refptr< webrtc::PeerConnectionFactoryInterface > factory)

public virtual ~PeerConnectionManager()

public virtual void sendSDP(PeerConnection* conn,const std::string & type,const std::string & sdp)

public virtual void sendCandidate(PeerConnection* conn,const std::string & mid,int mlineindex,const std::string & sdp)

public virtual void recvSDP(const std::string & peerid,const json::Value & data)

public virtual void recvCandidate(const std::string & peerid,const json::Value & data)

public virtual void onAddRemoteStream(PeerConnection* conn,webrtc::MediaStreamInterface * stream)

public virtual void onRemoveRemoteStream(PeerConnection* conn,webrtc::MediaStreamInterface * stream)

public virtual void onStable(PeerConnection* conn)

public virtual void onClosed(PeerConnection* conn)

public virtual void onFailure(PeerConnection* conn,const std::string & error)

public webrtc::PeerConnectionFactoryInterface * factory() const

protected rtc::scoped_refptr< webrtc::PeerConnectionFactoryInterface > _factory

class scy::StreamRecorder

class scy::StreamRecorder
  : public rtc::VideoSinkInterface< cricket::VideoFrame >
  : public AudioTrackSinkInterface

Summary

Members Descriptions
public StreamRecorder(constav::EncoderOptions& options)
public ~StreamRecorder()
public void setVideoTrack(webrtc::VideoTrackInterface * track)
public void setAudioTrack(webrtc::AudioTrackInterface * track)
public void onPacketEncoded(av::MediaPacket& packet)
public void OnFrame(const cricket::VideoFrame & frame) VideoSinkInterface implementation.
public void OnData(const void * audio_data,int bits_per_sample,int sample_rate,size_t number_of_channels,size_t number_of_frames) AudioTrackSinkInterface implementation.
protected av::MultiplexEncoder _encoder
protected rtc::scoped_refptr< webrtc::VideoTrackInterface > _videoTrack
protected rtc::scoped_refptr< webrtc::AudioTrackInterface > _audioTrack
protected bool _awaitingVideo
protected bool _awaitingAudio

Members

public StreamRecorder(constav::EncoderOptions& options)

public ~StreamRecorder()

public void setVideoTrack(webrtc::VideoTrackInterface * track)

public void setAudioTrack(webrtc::AudioTrackInterface * track)

public void onPacketEncoded(av::MediaPacket& packet)

public void OnFrame(const cricket::VideoFrame & frame)

VideoSinkInterface implementation.

public void OnData(const void * audio_data,int bits_per_sample,int sample_rate,size_t number_of_channels,size_t number_of_frames)

AudioTrackSinkInterface implementation.

protected av::MultiplexEncoder _encoder

protected rtc::scoped_refptr< webrtc::VideoTrackInterface > _videoTrack

protected rtc::scoped_refptr< webrtc::AudioTrackInterface > _audioTrack

protected bool _awaitingVideo

protected bool _awaitingAudio

class scy::VideoPacketSource

class scy::VideoPacketSource
  : public VideoCapturer

VideoPacketSource implements a simple cricket::VideoCapturer that gets decoded remote video frames from a local media channel. It's used as the remote video source's VideoCapturer so that the remote video can be used as a cricket::VideoCapturer and in that way a remote video stream can implement the MediaStreamSourceInterface.

Summary

Members Descriptions
public VideoPacketSource()
public virtual ~VideoPacketSource()
public virtual cricket::CaptureState Start(const cricket::VideoFormat & capture_format) cricket::VideoCapturer implementation.
public virtual void Stop()
public virtual bool IsRunning()
public virtual bool GetPreferredFourccs(std::vector< uint32_t > * fourccs)
public virtual bool GetBestCaptureFormat(const cricket::VideoFormat & desired,cricket::VideoFormat * best_format)
public virtual bool IsScreencast() const
public void onVideoCaptured(void * sender,av::VideoPacket& packet)

Members

public VideoPacketSource()

public virtual ~VideoPacketSource()

public virtual cricket::CaptureState Start(const cricket::VideoFormat & capture_format)

cricket::VideoCapturer implementation.

public virtual void Stop()

public virtual bool IsRunning()

public virtual bool GetPreferredFourccs(std::vector< uint32_t > * fourccs)

public virtual bool GetBestCaptureFormat(const cricket::VideoFormat & desired,cricket::VideoFormat * best_format)

public virtual bool IsScreencast() const

public void onVideoCaptured(void * sender,av::VideoPacket& packet)

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